[asterisk-bugs] [Asterisk 0015975]: Unable to change the packetization settings (ptime) for codecs from default of 20ms

[Asterisk Bug Tracker]

The following issue has been CLOSED 
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https://issues.asterisk.org/view.php?id=15975 
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Reported By:                jehanzeb
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15975
Category:                   Codecs/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           Older 1.4 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 open
Fixed in Version:           
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Date Submitted:             2009-09-28 05:27 CDT
Last Modified:              2009-10-07 08:58 CDT
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Summary:                    Unable to change the packetization settings (ptime)
for codecs from default of 20ms
Description: 
Hi, I am currently running Asterisk version 1.4.21
my problem is that even though i have tried to force outbound calls with a
codec packetization rate of 10ms, or 30ms, asterisk keeps sending the
Invite message with the default ptime of 20ms.

my sip.config file for this peer is 

[sylantro]
type=friend
disallow=all                    ; First disallow all codecs
disallow=gsm
allow=ulaw:10,alaw:30           ; Allow codecs in order of preference
autoframing=yes
context=testcontext
trustrpid = yes
;dtmfmode=inband
canreinvite=yes
host=195.219.133.219
port=5065

sip show peer command shows the following settings

Name       : sylantro
  Secret       : <Not set>
  MD5Secret    : <Not set>
  Context      : testcontext
  Subscr.Cont. : <Not set>
  Language     : en
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : No
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : no
  Nat          : RFC3581
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : Yes
  PromiscRedir : Yes
  User=Phone   : No
  Video Support: No
  Trust RPID   : Yes
  Send RPID    : Yes
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : auto
  LastMsg      : 0
  ToHost       : 195.219.133.219
  Addr->IP     : 195.219.133.219 Port 5065
  Defaddr->IP  : 0.0.0.0 Port 0
  Def. Username:
  SIP Options  : (none)
  Codecs       : 0xc (ulaw|alaw)
  Codec Order  : (ulaw:10,alaw:30)
  Auto-Framing:  Yes
  Status       : Unmonitored
  Useragent    :
  Reg. Contact :
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---------------------------------------------------------------------- 
 (0111975) lmadsen (administrator) - 2009-10-07 08:58
 https://issues.asterisk.org/view.php?id=15975#c111975 
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Aha. I think this is a dialplan configuration issue.

Try dialing it as:

Dial(SIP/${ARG1}@sylantro)

Or set the value of your global variable to 'sylantro'

Right now you're circumventing all the information you've applied to the
peer sylantro, and just dialing it via the IP address directly. That is why
setting it in the [general] section makes this work. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-10-07 08:58 lmadsen        Note Added: 0111975                          
2009-10-07 08:58 lmadsen        Status                   feedback => closed  
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